• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.989-993
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• 2008

본 연구에서는 생체신호 중 비침습적으로 측정이 가능하고 많은 건강정보를 포함하고 있는 ECG(electrocardiogram)신호를 일상생활 중 보다 편리하게 모니터링 할 수 있는 시스템을 구현하고자 하였다. 이를 위하여 벨트형 ECG전극 시스템을 개발하였으며, 배터리로 구동 가능한 초소형 저전력 ECG측정시스템을 구현하였다. 또한 측정된 ECG신호의 무선전송을 위하여 Zigbee호환 무선센서노드를 이용하여 초저전력 무선데이터 통신부를 구성하였고 PC상에서 ECG신호를 모니터링하기 위한 프로그램을 구현하였다. 그리고 ECG측정 시 움직임에 따라 발생하는 동잡음의 제거를 위하여 이동평균필터(moving average filter)를 이용하여 기저선 변화를 추출하였고 이를 적응필터의 참조신호로 사용하여 동잡음을 제거하였다. 실험 결과 본 연구에 의해 구현된 ECG전극 및 계측시스템을 통해 활동상태 에서도 ECG계측 가능성을 확인하였으며, 제안한 적응신호처리기법을 통해 활동 중 ECG측정에서 동잡음의 최소화가 가능함을 확인하였다.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.3
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• pp.224-229
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• 2009

In this paper, an efficient signal interference control technique to improve the convergence speed of Adaptive transversal filter with LMS algorithm is introduced. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, are analyzed to prove theoretically the improvement of convergence speed. According as the step-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Increasing the eigenvalue spread has the effect of controlling down the rate of convergence of the adaptive equalizer and also increasing the steady-state value of the average squared error and also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS Algorithms.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.2
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• pp.407-416
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• 2000

Adaptive digital filtering and noise cancelling technique using a tree structured filter bank are presented to reduce a undesirable aliasing due to the decimation of filtered output and improve the performance in terms of mean-square error and the convergence speed using a aliasing canceller. A signal is split into two subband by analysis filter bank and decimated by decimator and reconstructed by interpolation technique and synthesis filter bank. A variable step-size LMS algorithm is used to improve the convergence speed in case of existing the measurement noise in desired input of filter. It is shown by computer simulation that the proposed subband structure in this paper is superior to conventional subband filter structure in terms of mean-square error and convergence speed.

• Journal of Digital Convergence
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• v.10 no.11
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• pp.377-382
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• 2012

In this paper, the echo noise robust CHMM learning model using echo cancellation average estimator LMS algorithm is proposed. To be able to adapt to the changing echo noise. For improving the performance of a continuous speech recognition, CHMM models were constructed using echo noise cancellation average estimator LMS algorithm. As a results, SNR of speech obtained by removing Changing environment noise is improved as average 1.93dB, recognition rate improved as 2.1%.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.39 no.1
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• pp.76-84
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• 2002

In this paper, it is studied to remove the baseline wander and to minimize the distortion of ST segment in the noise filtering of ECG. In general, the standard filter and adaptive filter are used to remove the baseline wander of the ECG. But the standard filter is limited because the frequency of the baseline signal is variable and the apative filter is difficult to select the reference signal in case of using the adaptive filter. So we proposed a new method of the structure without reference signal using neural networks. To be convinced of the performance of this method, we used ECG data of MIT-BIHs. and obtained the result of the high performance,(-53.3[dB]) than standard filter(-16.3[dB]) and adaptive filter (-44.9[dB]).

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.2
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• pp.88-95
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• 1999

Stack filters are a class of digital nonlinear filters with excellent properties for signal restoration. Unfortunately, present algorithms for designing stack filters with large window size are limited in applications by their computational overhead and serial nature. In this paper, new, highly-parallel algorithm is developed for determining a stack filter which minimizes the mean absolute error criterion. It retains the iterative nature of the present adaptive algorithm, but significantly reduces the number of required to converge to an optima filter. A proof is also give that the proposed algorithm converges to an optimal stack filter.

• Transactions of the Korean Society of Mechanical Engineers
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• v.15 no.3
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• pp.809-823
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• 1991

본 연구에서는 Wiener 필터링 이론에 의하여 소음원의 입력신호에 대한 최적 한 예측을 할 수 있는 최적예측기(optimal predictor)로써 부가적인 음을 발생시키고 입력신호 및 출력신호 간의 차인 오차를 최소화시키도록 하는 적응신호처리방법에 대 해 설명하고 이러한 적응 신호처리 방법을 이용한 능동 소음 제어 방법을 제시하였다. 이와 아울러 제어계의 환경 변화에 따른 파라메타의 변화에 적응적으로 응답이 가능해 야 하는 적응 소음 제어계에서, 음향궤환과 함께 필히 고려해야하는 부가적인 전달함 수-모델과 스피커를 포함하는 보조경로 및 오차미이크로폰을 포함하는 오차경로의 전 달함수의 영향을 고려한 능동소음제어에 대해 연구하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.11
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• pp.1714-1725
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• 1993

This paper analyzes the properties of such algorithm that corresponds to the nonlinear adaptive algorithm with additional update terms, parameterized by the scalar factors a1, a2, a3 and Presents its structure. The analysis of convergence leads to eigenvalues of the transition matrix for the mean weight vector. Regions in which the algorithm becomes stable are demonstrated. The time constant is derived and the computational complexities of MLMS algorithms are compared with those of the conventional LMS, sign, LFG, and QFG algorithms. The properties of convergence in the mean square are analyzed and the expressions of the mean square recursion and the excess mean square error are derived. The necessary condition for the CFG algorithm to be stable is attained. In the computer simulation applied to the system identification the CFG algorithm has the more computation complexities but the faster convergence speed than LMS, LFG and QFG algorithms.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2013.06a
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• pp.351-353
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• 2013

본 논문에서는 현재 활발하게 표준화가 진행중인 3D-HEVC 의 기술들 중 Backward View Synthesis Prediction(BVSP)에서 움직임 보상을 수행하는 서브 블록 경계에 가우시안 필터를 적용하는 방법을 제안한다. BVSP 에서는 4x4 서브 블록 단위로 대표 깊이 정보를 구하여 움직임 보상을 수행하기 때문에 서브 블록 경계에 블록킹 왜곡이 발생할 수 있으므로 가우시안 필터를 통해 이러한 왜곡을 줄일 수 있다. 하지만 모든 경계 픽셀에 대해 가우시안 필터를 적용하지 않고 경계 픽셀의 주변 정보에 따라 적응적으로 가우시안 필터를 적용하고, 필터의 컨트롤 파라미터 또한 적응적으로 변경하는 방법을 제시한다. 제안하는 방법을 기존의 HTM 6.2 와 비교했을 때, 평균 0.1%의 부호화 효율 개선을 보이고 복잡도는 1.2% 증가 하였다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.74-80
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• 2008

Spatio-temporal filters are capable of reducing noise from moving pictures, which cannot be dealt with by spatial filters. However, the algorithm is too complicated to be realized as hardware. We implemented a real-time spatio-temporal noise reduction system, using at most three frames, based upon adaptive mean filter algorithm. Some factors which interfere with hardware implementation were modified. Noise estimated from the previous frame was used to filter the current frame so that filtering could be conducted in parallel with noise estimation. This speeds up the system thereby enabling real-time execution. The form of filtering windows was also modified to facilitate synchronization. The proposed structure was implemented on Virtex 4 XC4VLX60, occupying 66% of total slices with 80MHz of the maximum operation frequency.